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5e5d017c2b
Instead of getting header extension mapping from a receiver object, get the mapping from the received packet. The purpose is to be able to remove extension information from webrtc/call/receive_stream.h. Header extensions are negotiated per mid, not per receive stream. The goal is to reduce the number of places where packets are parsed and demuxed. Bug: webrtc:7135, webrtc:14795 Change-Id: I8944bc06a11dc572d9e14e7d7ee446a841096295 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288968 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38944}
71 lines
2.2 KiB
C++
71 lines
2.2 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "call/rtp_stream_receiver_controller.h"
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#include <memory>
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#include "rtc_base/logging.h"
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namespace webrtc {
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RtpStreamReceiverController::Receiver::Receiver(
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RtpStreamReceiverController* controller,
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uint32_t ssrc,
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RtpPacketSinkInterface* sink)
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: controller_(controller), sink_(sink) {
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const bool sink_added = controller_->AddSink(ssrc, sink_);
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if (!sink_added) {
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RTC_LOG(LS_ERROR)
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<< "RtpStreamReceiverController::Receiver::Receiver: Sink "
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"could not be added for SSRC="
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<< ssrc << ".";
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}
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}
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RtpStreamReceiverController::Receiver::~Receiver() {
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// This may fail, if corresponding AddSink in the constructor failed.
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controller_->RemoveSink(sink_);
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}
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RtpStreamReceiverController::RtpStreamReceiverController() {}
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RtpStreamReceiverController::~RtpStreamReceiverController() = default;
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std::unique_ptr<RtpStreamReceiverInterface>
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RtpStreamReceiverController::CreateReceiver(uint32_t ssrc,
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RtpPacketSinkInterface* sink) {
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return std::make_unique<Receiver>(this, ssrc, sink);
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}
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bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) {
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RTC_DCHECK_RUN_ON(&demuxer_sequence_);
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return demuxer_.OnRtpPacket(packet);
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}
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void RtpStreamReceiverController::OnRecoveredPacket(
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const RtpPacketReceived& packet) {
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RTC_DCHECK_RUN_ON(&demuxer_sequence_);
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demuxer_.OnRtpPacket(packet);
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}
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bool RtpStreamReceiverController::AddSink(uint32_t ssrc,
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RtpPacketSinkInterface* sink) {
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RTC_DCHECK_RUN_ON(&demuxer_sequence_);
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return demuxer_.AddSink(ssrc, sink);
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}
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bool RtpStreamReceiverController::RemoveSink(
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const RtpPacketSinkInterface* sink) {
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RTC_DCHECK_RUN_ON(&demuxer_sequence_);
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return demuxer_.RemoveSink(sink);
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}
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} // namespace webrtc
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